Monday May 31 2004 |
Time | Replies | Subject |
11:59PM |
3 |
G.729 fallback |
10:34PM |
1 |
Failover: iconnecthere to voicepulse |
8:47PM |
1 |
zapras how to |
7:29PM |
2 |
Billing and CDR's |
7:07PM |
1 |
Asterisk and SER Setup Questions. |
6:11PM |
2 |
scandsp, voicetronix and rxfx |
5:22PM |
0 |
Fedora Core Soft Phone that works? |
5:03PM |
1 |
audio problems between asterisk and Cisco 7910 using SCCP |
4:18PM |
4 |
wake-up call |
3:13PM |
0 |
Fwd: [Serusers] CDR mediation for VoIP |
3:04PM |
1 |
Firefly / LibIAX2 |
2:56PM |
0 |
digium card fax detect AND spandsp |
2:51PM |
0 |
MGCP RFC 3015 |
1:39PM |
0 |
SIP auth in/outside of nat? |
12:53PM |
0 |
Import Master.csv in the cdr_mysql database |
10:02AM |
1 |
line config |
9:38AM |
1 |
I want to purchase atleast one used quicknet card |
9:07AM |
2 |
Crc4 issues |
8:23AM |
0 |
Disclaimer fax number? |
8:11AM |
1 |
Chan Capi Audio Quality Issue... |
7:44AM |
5 |
D-Channel Problems |
7:08AM |
1 |
Where is my normal dialtone? With DLINK DG-104S (MGCP) |
6:54AM |
1 |
* on Opteron |
6:21AM |
2 |
Meetme + Billing |
6:15AM |
6 |
Asterisk and Zaptel for 2.6 kernel |
6:12AM |
2 |
Users in MySQL |
5:32AM |
1 |
(no subject) |
3:48AM |
1 |
Updated Zaptel this morning and *BOOM* *CRASH* |
2:51AM |
3 |
Quicknet PhoneJack Configuration |
1:49AM |
4 |
Need guides on setting up PDA on asterisk server |
12:59AM |
1 |
Quicknet PhoneJack Configuration files |
|
Sunday May 30 2004 |
Time | Replies | Subject |
8:08PM |
3 |
spandsp w/libtiff-3.6.1? |
6:05PM |
1 |
Advanced access control and cause handling |
6:01PM |
11 |
New Firefly version |
5:48PM |
1 |
Fwd: hdlc and kernel 2.4.26 |
3:41PM |
0 |
Codec details in CDR? |
12:35PM |
5 |
Unblocking incoming SIP |
12:01PM |
4 |
Sipura-spa2000 |
9:47AM |
2 |
Distribution of Linux |
8:00AM |
1 |
G729 Beta Codec |
5:48AM |
0 |
bri-stuff.0.0.2 |
5:41AM |
6 |
Compiling Asterisk with gcc 3.4 |
2:32AM |
6 |
*** Asterisk Sunday News: Gone Fishing... |
1:47AM |
1 |
MacOS X softphone IAX clients? |
|
Saturday May 29 2004 |
Time | Replies | Subject |
9:51PM |
2 |
Snom and multiple lines |
8:32PM |
0 |
E164.org Updates |
6:49PM |
1 |
Webmin Module in download directory |
6:42PM |
0 |
Asterisk - Zaptel - DIGIUM x 4 T1 |
6:00PM |
1 |
iConnectHere broken? |
4:01PM |
5 |
extracting country code from a number |
12:25PM |
1 |
SIP extension |
11:12AM |
3 |
Odd behaviour with "asterisk -rx" |
11:04AM |
4 |
PlayTones problem |
9:01AM |
1 |
Delay when routing PSTN -> IAXy dect phone |
7:45AM |
2 |
Caller ID with DID |
6:16AM |
1 |
transfer bug (#701 -> remote party hears alison, not me) |
12:49AM |
1 |
Example: Caller*ID Fixup Macro for use with DIDs |
|
Friday May 28 2004 |
Time | Replies | Subject |
10:55PM |
0 |
PacketCable with asterisk? |
9:20PM |
0 |
HDLC kernel 2.4.26 |
6:20PM |
0 |
Problem with digits blending on inbound puls ed digits? |
4:44PM |
2 |
Sipura SPA-3000 reviews? |
2:50PM |
0 |
Problem with digits blending on inbound pulsed digits? |
2:16PM |
1 |
TDM31B and Zaptel: FXO port not recognized? |
2:11PM |
0 |
Help ! Echo on sip call. |
2:11PM |
0 |
Not call pickup for call to sip from mgcp phone |
1:48PM |
1 |
[Fwd: Re: call pickup fails.] |
1:08PM |
6 |
Beep Sound |
12:04PM |
0 |
SIP 404 error.... |
11:57AM |
3 |
Disable blind xfer |
11:51AM |
4 |
PCI 2.2 ?? |
11:47AM |
1 |
* will not load, after latest CVS install |
11:13AM |
1 |
[is this appropriate?] Anyone work for voicepulse or have experience with them |
9:43AM |
0 |
seeking an example for Message Waiting Indicator stutter dialtone |
9:32AM |
16 |
Asterisk Receptionist manager program. |
9:29AM |
0 |
sip client can dial with no registration |
9:24AM |
0 |
memory error? TE405P problem? |
9:17AM |
1 |
Zap callgroup/pickupgroup question |
8:52AM |
1 |
Fw: Asterisk and MySQL |
8:31AM |
1 |
Problems with PPP internet T1 |
8:10AM |
11 |
Asterisk Database |
7:43AM |
0 |
E1 channel bank problem |
7:17AM |
5 |
Time to lock down v1.1? |
6:34AM |
9 |
* as pri_net? |
6:27AM |
2 |
spandsp wont compile. |
6:13AM |
5 |
Asterisk and MySQL |
6:11AM |
0 |
FreeBSD admins * PLEASE HELP! |
5:52AM |
0 |
No Sound Card and No Sound from Phone |
5:18AM |
1 |
Immortal SIP & NAT problem |
5:09AM |
0 |
INTERTEX AND ASTERISK |
5:08AM |
1 |
JTAPI Interface in Asterisk |
4:44AM |
0 |
Call transfering |
3:52AM |
4 |
Wiki TOS - worrying for an open source project? |
3:19AM |
5 |
SIP Changes??? |
2:49AM |
1 |
asterisk console messages |
2:37AM |
2 |
Asterisk with Draytek 2600V |
2:16AM |
3 |
Asterisk addons |
1:48AM |
3 |
2 Avm fritz passive card in the same box |
1:19AM |
0 |
dialogic was RE: "Glare" condition - How well does asteriskhandle? |
|
Thursday May 27 2004 |
Time | Replies | Subject |
9:41PM |
0 |
mysql-vm-routines does not use the context properly |
8:57PM |
0 |
Hangup problem during intergration with 3rd party pbx |
8:28PM |
1 |
New to Asterisk - 2 question |
8:22PM |
0 |
No stutter MWI on zaptel channel with message waiting |
8:11PM |
0 |
Asterisk RPMS Updated (0.9.0 for RH73,8,9 and FC1) |
7:36PM |
0 |
HFC-S BRI Slack9.1 kernel 2.6.6 "Guide" bri-stuff.0.0.2 |
4:50PM |
0 |
[OT] spandsp hylafax asterisk and confusion |
2:53PM |
0 |
seeking H.323 <-> MGCP (User Agent) gateway |
2:32PM |
4 |
AGI Pascal |
2:25PM |
0 |
Re: Asterisk-Users digest, Vol 1 #3921 - 15 msgs |
2:21PM |
0 |
Billing, Radius, anyone? |
1:26PM |
2 |
Scroll mode in cli |
1:01PM |
1 |
Holding and call parking idiosyncrasies... |
12:45PM |
0 |
Cisco 7940/60 sip downloads |
12:26PM |
5 |
Silly incoming SIP failure |
11:20AM |
0 |
Zaptel, analog phone, and call waiting |
11:12AM |
0 |
400 Bad request?? |
10:59AM |
1 |
cvs problem with TDM04B ? |
10:11AM |
1 |
Dlink DG-104s telnet reboot |
10:07AM |
0 |
WIKI voip-info.org up again |
8:55AM |
1 |
Queue Hold Time |
8:53AM |
3 |
generate dial tone |
8:21AM |
2 |
Asterisk and PostgreSQL |
7:44AM |
0 |
Save voice data from IAX. Is that posible? |
7:32AM |
5 |
FireFly doesn't work with 3rd party anymore |
7:04AM |
0 |
zaphfc: All DTMF tones are doubled |
6:32AM |
3 |
dialogic was RE: "Glare" condition - How well does asteriskhandle? |
6:30AM |
0 |
threewaycalling |
6:04AM |
0 |
return from call parking... |
6:02AM |
1 |
What a Difference a NIC makes |
5:58AM |
1 |
opinions on oneunified.net as asterisk provider |
5:49AM |
6 |
CAPI / Channels |
5:44AM |
4 |
Wiki down |
5:43AM |
0 |
Simple call generator (shell script) |
4:54AM |
0 |
asterisk-oh323, new version 0.6.2 |
4:46AM |
0 |
bri-stuff-0.0.2 ported to actual development CVS (head) |
3:20AM |
0 |
Does anyone know a manager-phone working with asterisk |
2:49AM |
0 |
Changing IAX2 refresh times |
1:27AM |
2 |
Freenet iPhone w/Asterisk |
1:01AM |
1 |
Astersik and PostgreSQL |
12:00AM |
1 |
call pickup fails. |
|
Wednesday May 26 2004 |
Time | Replies | Subject |
11:41PM |
1 |
dialplan AGI DTMF |
11:14PM |
2 |
Voicetronix OpenLine4 -- Help Needed |
10:57PM |
0 |
quicknet jack card |
9:18PM |
1 |
BudgeTone IP phone in Hong Kong |
8:13PM |
2 |
Help! No stutter dialtone on message waiting - zaptel phones |
5:55PM |
0 |
modprobe wcfxo hangs server |
4:39PM |
0 |
Echo on sip to sip call |
4:38PM |
2 |
VOIP Service Providers |
4:26PM |
1 |
ztdummy with kernel 2.6 |
2:18PM |
1 |
sip_reg_timeout problem |
2:17PM |
0 |
Voicemail Recordings |
1:33PM |
1 |
intercept ringing phone |
12:36PM |
0 |
Asterisk / SER or both |
12:11PM |
4 |
Sipura stun settings |
12:06PM |
2 |
SPAM MESSAGE - [Asterisk-Dev] warning message (sound card) - when I run asterisk!!! |
11:11AM |
2 |
tieline digit timeout |
11:02AM |
2 |
Monitoring Calls |
9:30AM |
2 |
tdm04b stopped taking inbound calls - todays cvs |
9:26AM |
1 |
Rejecting Calls (SIT Tone/Invalid) Across PRI |
8:47AM |
0 |
Newbie Here |
7:51AM |
1 |
outgoing MSN on zaphfc |
7:33AM |
3 |
bug or feature? |
6:40AM |
7 |
PostgreSQL |
6:37AM |
2 |
Anyone got latest SIP image for Cisco 7960? |
6:26AM |
0 |
Sound Distortion using IAX? |
5:13AM |
0 |
SV: dialing multiple extensions |
2:34AM |
15 |
* INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW * |
2:27AM |
0 |
act via Network or web? |
1:22AM |
5 |
cdr_odbc with mysql on a remote server |
12:58AM |
9 |
CTI (Computer-Telephony Integration) with Asterisk ? |
12:49AM |
0 |
NMI occures while loading the zaptel module |
12:35AM |
1 |
ISDN BRI Problem |
12:28AM |
4 |
Forwarding and record |
|
Tuesday May 25 2004 |
Time | Replies | Subject |
9:42PM |
4 |
Can I do this ... |
8:36PM |
1 |
CVS checkout problem |
3:43PM |
1 |
problem with vigor 2600v |
3:39PM |
0 |
No sound for MusicOnHold and SayDigits |
3:37PM |
1 |
No ringing on inbound DID calls |
3:27PM |
3 |
"Glare" condition - How well does asteriskhandle? |
2:16PM |
1 |
D-Channel on span 1 up/down + frame slips with zaptelBRI |
2:07PM |
0 |
MSN selection when dialout ISDN (ttyI* modem -interface, NOT CAPI) |
1:01PM |
1 |
SMS |
12:52PM |
6 |
Downgrading Asterisk |
12:47PM |
3 |
Voice Pulse |
11:53AM |
8 |
"Glare" condition - How well does asterisk handle? |
11:46AM |
1 |
FYI: Cisco firmware 7.1 released |
11:35AM |
4 |
79XX converting |
11:25AM |
0 |
Still Adtran and T100p |
11:01AM |
1 |
Call Admission Control |
10:56AM |
3 |
Telus: Overseas calling |
10:48AM |
0 |
asterisk and partner acs |
10:40AM |
1 |
Unable to create channel of type 'CAPI' |
9:14AM |
1 |
voicemail notify to external number |
8:38AM |
1 |
high pitched tone and message on answer |
8:17AM |
1 |
using asterisk with iptel addreses |
7:51AM |
1 |
(no subject) |
7:23AM |
1 |
dialout=fromvm |
7:08AM |
1 |
Zhone Zplex issues |
6:48AM |
0 |
Problems with IAX configuration |
6:29AM |
1 |
Problem - Adtran TSU 600, t100p |
6:27AM |
0 |
SMDi Interface? |
6:13AM |
4 |
fax/sandsp segfaulting asterisk |
5:56AM |
10 |
spandsp hylafax asterisk and confusion |
4:51AM |
1 |
SS7 links |
4:38AM |
0 |
Asterisk and Sipp |
4:08AM |
0 |
Compiling on OSX 10.3.3 |
3:44AM |
1 |
Troubles with Kphone] |
3:43AM |
0 |
[Fwd: Answer App hanging in I4L] |
3:30AM |
1 |
Answer App hanging in I4L |
3:15AM |
0 |
Sound card problem |
2:52AM |
1 |
Troubles with Kphone |
2:21AM |
2 |
sip phone problem |
2:07AM |
4 |
Sip/IAX Clients for Linux |
1:50AM |
0 |
Question IAX and SIP bound to different IP's on the same * box |
1:31AM |
1 |
Speed Dials |
1:21AM |
1 |
SipTone II and Choppy/Stuttering Audio |
12:20AM |
1 |
Using Ser and Asterisk together |
|
Monday May 24 2004 |
Time | Replies | Subject |
9:37PM |
1 |
threewaycalling and # |
7:43PM |
1 |
no delivery from queue on IAX2 extension |
7:18PM |
6 |
11 instead of Star |
7:13PM |
2 |
Newbie extensions.conf I need to include [SMS] context. |
5:20PM |
3 |
100 analog phones?? HOWTO? |
3:26PM |
4 |
Grandstream message light button |
3:11PM |
5 |
mpg123 |
2:58PM |
0 |
Asterisk connected to DataBase |
2:55PM |
1 |
Using Blacklist |
2:25PM |
2 |
SIP Authentication Problem |
2:12PM |
7 |
Sip Registration Problem |
1:44PM |
0 |
H.323, video and asterisk.... |
1:15PM |
0 |
NCS support? |
1:08PM |
4 |
using the asterisk mailbox utility |
1:00PM |
2 |
testing asterisk on FXS lines |
12:29PM |
1 |
Re: Asterisk-Users digest, Vol 1 #3883 - 13 msgs |
12:12PM |
2 |
TerraCall Setting |
12:00PM |
1 |
CDR destination when user presses '#' |
11:47AM |
4 |
dialing multiple extensions |
11:13AM |
1 |
extensions/sip from database? |
10:34AM |
1 |
Channelized T1, SIP phones, HW Echo Canceller |
10:19AM |
2 |
IP local loop? |
9:07AM |
0 |
Help with IAX , voice Distortion or Breakage |
8:41AM |
1 |
Remote Disconnects |
8:31AM |
0 |
zapata ? question |
8:21AM |
3 |
Meetme Options (new one) |
8:17AM |
1 |
Fw: creating a single user voice mail box on asterisk? |
8:17AM |
1 |
Fw: setting the number of rings befor asterisk picks up? |
7:35AM |
0 |
ZapRas problems |
7:17AM |
1 |
Chan_capi 0.3.1 , Asterisk , 3 x C4 active ISDN card Segmentation fault |
7:07AM |
1 |
detecting pick up? |
6:32AM |
1 |
zaprtc-for-2.6 |
6:27AM |
3 |
Help with IAX , voice Distortion or Breakage. |
6:13AM |
0 |
Variable AGI Parser |
5:53AM |
1 |
Cisco & Asterisk |
4:58AM |
1 |
SetVar - bellcode and cisco phone |
4:00AM |
0 |
IAX problems using CVS HEAD, but not CVS STABLE |
2:03AM |
0 |
Asterisk Audio Problem |
12:54AM |
0 |
updating the dial application |
12:29AM |
5 |
2 Sip phones behind un-natted Asterisk |
12:19AM |
1 |
Problem receiving a fax with RxFAX |
|
Sunday May 23 2004 |
Time | Replies | Subject |
11:40PM |
0 |
STREAM FILE question |
10:46PM |
2 |
Document |
9:49PM |
1 |
Aastra ADSI phone |
7:52PM |
5 |
PRI problem??? |
7:36PM |
1 |
asterisk prompts? |
6:19PM |
0 |
setting the number of rings befor asterisk picks up? |
6:05PM |
0 |
Sipura SPA-3000 Beta |
5:48PM |
0 |
HELP!!! How do I move voicemail files to a new machine? |
5:39PM |
1 |
Serious NAT problems: can't call between lines on sipura |
5:08PM |
0 |
Fwd: regulating voip - aca |
4:37PM |
3 |
NetJet and RAS |
4:10PM |
4 |
Asterisk Prepaid |
3:56PM |
0 |
creating a single user voice mail box on asterisk? |
12:45PM |
0 |
ztdummy - how to test? |
12:09PM |
1 |
extension pattern matching |
8:05AM |
1 |
IAX2 NAT / Registration Issue |
7:57AM |
1 |
ZAPTEL not loading on FC2 |
6:40AM |
1 |
IAX2 REACHABLE/UNREACHABLE |
5:22AM |
1 |
*** Asterisk Sunday News: Conferences on the phone and IRL - "in real life" |
4:11AM |
1 |
SIP with TerraCall Error |
|
Saturday May 22 2004 |
Time | Replies | Subject |
11:21PM |
1 |
Sip proxy registration help |
7:45PM |
0 |
CallerID and AON in Eastern Europe |
7:40PM |
1 |
Problems using Adtran 750 FXO and TE405P |
6:24PM |
1 |
Asterisk-oh323 0.6.1 Compiling problem |
5:40PM |
3 |
e164.org |
4:16PM |
4 |
sip call using name in sip.conf |
3:43PM |
1 |
app_queue and app_groupcount |
3:38PM |
0 |
ast_rtp_read: Unknown RTP codec 72 received |
3:24PM |
14 |
Caller ID with BT CD50 |
3:11PM |
5 |
Asterisk firewall config |
2:50PM |
0 |
HOW do I restore voicemail from backups? |
1:44PM |
1 |
Failure while compiling |
12:32PM |
1 |
Re: Sipura and STUN (was: rejected NOTIFY re quests) |
11:56AM |
1 |
Re: Sipura and STUN (was: rejected NOTIFY requests) |
10:57AM |
2 |
loader.c:240 ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_moh_stop |
9:51AM |
2 |
RxFAX generates no tiff file |
9:49AM |
3 |
fwd on busy when calling multiple extensions at once |
9:34AM |
0 |
asterisk cpu load |
8:39AM |
2 |
How to share Zap channels in 2 Asterisk servers |
7:36AM |
1 |
Dynamic SIP.CONF |
7:18AM |
2 |
rejected NOTIFY requests |
5:36AM |
1 |
call waiting indicator do not work for me. |
3:24AM |
2 |
Chan CAPI and Latest CVS wont compile |
12:44AM |
1 |
Is it Possible |
|
Friday May 21 2004 |
Time | Replies | Subject |
7:29PM |
4 |
G.729a beta codec on old Pentiums |
6:39PM |
6 |
VoicePulse SIP |
5:21PM |
1 |
Dumb TDM400P question |
4:25PM |
5 |
T100P HDLC configuration |
3:25PM |
4 |
dial application - continue in context |
1:57PM |
5 |
Making a SIP call |
1:51PM |
0 |
voicemail removal script |
1:20PM |
2 |
Asterisk upgrade on production box |
11:07AM |
0 |
Bridge calls |
10:48AM |
3 |
Asterisk and OH323 |
10:01AM |
1 |
SIP Link on Web Pages |
9:57AM |
2 |
Failed to bind to 0.0.0.0:5060: Address already in use |
6:20AM |
0 |
Re: Your details |
4:46AM |
2 |
dial an IP address |
1:40AM |
4 |
Some problems with download Asterisk-addons |
1:25AM |
0 |
unable to use EXEC in AGI |
|
Thursday May 20 2004 |
Time | Replies | Subject |
9:10PM |
2 |
SwissVoice IP phones |
8:21PM |
3 |
UIP 200 |
5:19PM |
0 |
Re: need phones with line and call indicators |
4:47PM |
4 |
Problem with SIP softphone |
4:39PM |
0 |
Variable value lost after call parking |
3:46PM |
8 |
I have put iLBC at the top |
3:43PM |
0 |
MSSQL2000 + cdr_odbc.c fix (WAS: problem with cdr_odbc) |
2:42PM |
0 |
Phone_read Resource temporarily unavailable |
1:47PM |
0 |
rtpmap issue w/Grandstreams |
11:26AM |
1 |
G729A problem |
10:50AM |
1 |
Avaya Partner Phones to SIP? |
10:33AM |
0 |
Re: My details |
10:28AM |
4 |
snom 200 and hold |
10:28AM |
3 |
Anonymous sip register |
10:08AM |
0 |
Error running festival command |
10:04AM |
2 |
asterisk-providers mailing list? |
9:52AM |
6 |
VoicePulse broken? |
9:20AM |
0 |
Time Limit Warning File |
9:16AM |
2 |
codec used on E1 |
8:58AM |
6 |
G729 codec for asterisk |
8:57AM |
1 |
Premisys Slimline CB |
8:40AM |
4 |
Mystery SIP channels |
8:30AM |
0 |
budgetone problem on hangup |
8:15AM |
1 |
FC2 compile of zaptel |
8:07AM |
1 |
Mysql |
7:54AM |
1 |
Tellabs 2572 Configuration Advice? |
7:44AM |
4 |
x100p card + dailing out |
6:39AM |
1 |
Grandstream tftp cfg.txt format |
6:18AM |
1 |
Softphone Audio problem |
5:47AM |
1 |
voicemail customization |
5:19AM |
0 |
MultiTech MVP200 and Iconnect |
4:59AM |
1 |
Problems with Quadbri card |
4:14AM |
2 |
AGI/php script not working |
2:57AM |
0 |
DTMF problems to connect CME to Asterisk. |
2:41AM |
0 |
disa issue |
2:12AM |
2 |
Fedora Core 2 and Kernel 2.6 |
1:20AM |
2 |
Softphone lag |
|
Wednesday May 19 2004 |
Time | Replies | Subject |
10:19PM |
1 |
Using stutter dialtone like the PSTN does |
9:58PM |
1 |
asterisk - cisco as5400 (h,323) - PSTN |
7:42PM |
2 |
MGCP error dialing |
6:57PM |
2 |
O'Reilly Open Source Convention in Portland |
6:30PM |
1 |
avoiding rtp triangle |
4:48PM |
2 |
CallCenter setup |
1:52PM |
1 |
voicemail notify problem on sip extension |
1:41PM |
0 |
problem with ignorepat |
1:14PM |
3 |
Remote Call Forwarding |
12:58PM |
1 |
I'm in ADSI hell... |
12:46PM |
2 |
how to pass call duration to an agi script |
12:46PM |
1 |
Swissvoice ip10: No 3-way-calling! (MGCP) |
12:40PM |
3 |
Call recording between SIP phones |
12:19PM |
0 |
H323 + asterisk + NETMEETING |
11:46AM |
1 |
One-way audio with H.323 --> SIP call |
11:39AM |
1 |
iconnect register problem |
11:21AM |
0 |
example of mulity company extension.conf needed. |
10:51AM |
4 |
TDM400P problems with 1 FXS, 1 FXO |
9:52AM |
1 |
xp100 not hanging up after call disconnect |
9:44AM |
1 |
Strange Sip (FWD, SipGate and such) problem |
9:37AM |
4 |
Asterisk External Ringing |
7:49AM |
1 |
Gotta love Ellen Muraskin (RE: OMG THE SKY IS FALLING!! NOT!!!) |
6:55AM |
0 |
Video support SIP and IAX2 |
5:58AM |
2 |
persistant call variables |
4:42AM |
1 |
Old sound in new call. |
3:41AM |
1 |
verify Request URI |
2:33AM |
1 |
What has happened to my asterisk/PRI ? |
2:12AM |
1 |
FreeBSD + Zaptel + Asterisk |
1:47AM |
1 |
using iLBC |
1:45AM |
2 |
Problem compiling zaptel with BRIstuff 0.0.2 |
|
Tuesday May 18 2004 |
Time | Replies | Subject |
10:08PM |
1 |
Linejack dialout |
9:26PM |
5 |
want to set a var in sip.conf |
9:06PM |
0 |
FW: * and Cisco routers |
8:55PM |
0 |
Asterisk not answering phone |
8:23PM |
0 |
Asterisk to IAXTel help |
6:54PM |
3 |
Free Softphone Recomendations |
6:42PM |
7 |
Asterisk on Compact PCI platform |
6:19PM |
3 |
call announce? in MeetMe? |
5:50PM |
2 |
My TDM-400P FXO experience |
5:45PM |
2 |
* and Cisco routers |
5:05PM |
1 |
Problem with QuadBRI |
5:04PM |
2 |
ADIT 600 Manual |
4:48PM |
0 |
tying a call to indications/call progress tones |
3:00PM |
0 |
using ast_request("zap", format, "pseudo")? |
2:09PM |
0 |
zaphfc Compile Error |
1:31PM |
1 |
how does a sip://user@dom.ain url come in |
12:50PM |
1 |
Asterisk on OS X |
12:48PM |
0 |
snom 200 phones. |
12:35PM |
0 |
MeetMe conference delay increasing |
12:18PM |
1 |
TDM400P and AGGRESSIVE_SUPPRESSOR dropping calls |
12:18PM |
5 |
blocked caller id |
11:28AM |
5 |
AArgh, * and the 7960 |
10:38AM |
0 |
PRI CID name |
10:06AM |
0 |
403 Forbidden since upgrading |
9:01AM |
1 |
How can I dial (0 + telephone number) |
8:45AM |
11 |
ATA devices |
8:23AM |
2 |
registering in sipphone |
8:13AM |
5 |
X100P answer in first Ring |
7:43AM |
1 |
G.729 on /dev/sda |
7:24AM |
2 |
problem with cdr_odbc |
7:19AM |
1 |
VoiceMailMain dumps user back into my incoming context after leaving a message |
7:04AM |
1 |
VoIP Termination w/ 402 or 712 area code? |
6:41AM |
2 |
asterisk voicemail retrieval using a cisco 7940 |
6:40AM |
0 |
oh323.conf |
6:27AM |
0 |
problems with asterisk-oh323 |
4:46AM |
1 |
R: Configure asterisk for outgoing.. need authuser parameter? |
4:46AM |
1 |
Dial and MeetMe on the same channel |
4:19AM |
0 |
No luck using asterisk as proxy... |
4:03AM |
1 |
Configure asterisk for outgoing.. need authuser parameter? |
3:49AM |
3 |
Q.931 clearing causes |
2:50AM |
1 |
DateTime bug? |
2:04AM |
0 |
Outbound call using Soft Phone |
1:47AM |
0 |
Number portability |
12:35AM |
0 |
Re: X100P Red Alarm Ireland |
|
Monday May 17 2004 |
Time | Replies | Subject |
8:59PM |
2 |
Problems w. chan_capi + ztdummy |
8:27PM |
0 |
failed compile |
8:20PM |
0 |
Corrupt Callerid Data |
5:53PM |
0 |
mgcp with busy tone |
5:52PM |
0 |
CAPI<->SIP broken incoming audio |
5:20PM |
0 |
*, Sipura, Call-Waiting, X100P, 2 ZAP Calls |
5:13PM |
0 |
Rate Engine Application |
3:59PM |
4 |
total newbie sanity check |
3:07PM |
1 |
problems compiling h323 support |
2:33PM |
0 |
Something weird |
2:29PM |
0 |
DTMF transmitted over IAX2 coming out as clicks at the other end |
1:56PM |
0 |
RE: question on domains requiring SRV lookups within asterisk |
1:12PM |
0 |
Astricon 2004 - the developer's meeting ** CALL FOR PAPERS |
1:09PM |
0 |
iax2 and ethereal |
12:49PM |
0 |
Zap callwaiting hookflash idiosyncracy/flaw? |
11:51AM |
4 |
*8 problem still there? |
10:27AM |
0 |
dial without answer ? |
9:41AM |
0 |
span_dsp & faxing: segmentation fault |
9:35AM |
4 |
Redhat 7.3 compiling problem |
8:54AM |
0 |
Snom200 Firmware: I only see 2.04g |
8:41AM |
1 |
Snom200 Firmware: I only see 2.04g |
7:45AM |
0 |
openbsd compilation fails for recent checkout of v1-0_stable |
7:17AM |
0 |
Cisco 7940/7960 users |
6:10AM |
0 |
Some thougts about implementing native 3-way calling and attended transfer |
5:38AM |
2 |
recommended hardware for quad E1 system |
2:28AM |
4 |
Asterisk Proxy Type |
1:59AM |
0 |
Tones... |
12:52AM |
2 |
Grandstream phone from speaker phone back to handset |
|
Sunday May 16 2004 |
Time | Replies | Subject |
7:20PM |
2 |
Re: say.c compilation error |
7:08PM |
0 |
Caller ID from Call Pickup |
6:39PM |
2 |
(no subject) |
2:17PM |
4 |
X100P Ireland Red Alarm (AR Tarzi) |
1:06PM |
0 |
CallerID information on H.323 channel |
12:15PM |
6 |
X100P problem with PSTN from BOLIVIA |
9:51AM |
1 |
Vertical applications? |
9:03AM |
0 |
Call forking/parallel call cdr. |
7:21AM |
2 |
cdr-csv / Zaptel BRI / full number not stored (missing leading zeroes) |
6:31AM |
2 |
Snom200 ? |
6:07AM |
7 |
Grandstream v1.0.4.68 firmware |
3:37AM |
1 |
** Asterisk Sunday Morning News: Contribute to the community |
12:34AM |
0 |
sip show channels |
12:32AM |
0 |
snoom200 and g729 |
|
Saturday May 15 2004 |
Time | Replies | Subject |
10:50PM |
1 |
TDMoE hangs the machine |
10:02PM |
0 |
*8 (call pickup) using Manager or AGI interfaces? |
7:43PM |
1 |
110 extensions |
7:11PM |
1 |
Newbie question-no outgoing audio |
4:20PM |
4 |
Dial to Answer -- Can this be done? |
3:21PM |
0 |
Some doc |
12:01PM |
2 |
Subject: Re: X100P Ireland Red Alarm |
11:43AM |
1 |
Voicemail transfer |
9:47AM |
1 |
G729 Registration unsuccessful |
9:13AM |
1 |
Power alarm on module 1, resetting. |
8:07AM |
0 |
echocancelwhenbridged=no ? |
5:15AM |
1 |
asterisk with E1 |
1:17AM |
1 |
X100P Ireland Red Alarm |
|
Friday May 14 2004 |
Time | Replies | Subject |
11:26PM |
4 |
IP-PSTN / PSTN-IP Gateway Service Providers |
9:40PM |
2 |
Data through T1, nethdlc |
9:00PM |
1 |
Dead FXO Module on TDM400P? |
7:14PM |
0 |
MGCP information |
6:03PM |
0 |
Bandwidth measurement tools (was: GSM v iLBC for low bandwidth connections) |
5:04PM |
4 |
Nufone.net? |
3:46PM |
3 |
SoftPhone to SoftPhone with No Voice |
2:36PM |
0 |
Interested on Sip to PSTN gateway |
2:28PM |
0 |
[Daniel Golding] Re: New VOIP Peering/Interconnection Mailing List Announcement |
2:12PM |
1 |
Loop length supported by FXS module on Wildcat TDM400 card |
1:22PM |
0 |
SuSE & Zaptel Compilation errors |
12:32PM |
3 |
X100P and TDM400P non-USA Caller ID |
11:57AM |
2 |
Fwd: [ISN] Voice Over IP Can Be Vulnerable To Hackers, Too |
11:47AM |
7 |
What's in ${EXTEN} ? Why does voicemail prompt for an extension? |
11:47AM |
4 |
How to Echo extension number to caller? |
11:43AM |
4 |
app_dbmysql and ODBC Voicemail |
10:37AM |
2 |
Scalable IVR |
10:36AM |
0 |
"skipping" / dropped audio, high-pitched squeal on X100P channels.. |
9:10AM |
2 |
grandstream and timeservers |
8:53AM |
7 |
Business Looking Analog Phone |
8:30AM |
0 |
vovidia |
8:28AM |
4 |
snom 2.05b firmware |
8:24AM |
5 |
OMG THE SKY IS FALLING!! NOT!!! |
8:07AM |
0 |
Overriding indications for IAX2 calls |
7:44AM |
1 |
Caller ID with NAME on PRI |
6:50AM |
0 |
TDM40P |
4:31AM |
2 |
GSM v iLBC for low bandwidth connections |
4:25AM |
0 |
Large Scale Deployments |
3:51AM |
0 |
Perhaps France isn't asleep also or a least part of it. |
3:11AM |
3 |
snom & gsm codec |
2:53AM |
2 |
Help needed with bri-stuff.0.02. slw91 k2.6.5 |
2:19AM |
3 |
Psssst. The US is asleep - let's talk intern ationalization !!! |
1:29AM |
0 |
3 Q's about queues and agents |
1:29AM |
4 |
snom200 call wait indication |
1:06AM |
1 |
chan_capi broken incoming audio |
12:26AM |
1 |
Psssst. The US is asleep - let's talk internationalization !!! |
12:10AM |
4 |
sip authentication |
|
Thursday May 13 2004 |
Time | Replies | Subject |
9:25PM |
0 |
NI2 signaling with * for CID&NAME? |
7:36PM |
2 |
Asterisk, Configuration of SDP in SIP messages |
5:16PM |
0 |
Consult transfer on SNOM 105 |
4:38PM |
1 |
pattern matching w/ Cisco dialplans |
3:36PM |
1 |
TDM11B - 2nd channel is not detected |
3:09PM |
0 |
(no subject) |
2:57PM |
1 |
MeetMe with AGI scripts |
2:44PM |
0 |
vocera |
2:43PM |
3 |
recommend a Linux based TFTP server |
2:24PM |
6 |
IAXy |
1:40PM |
4 |
IAX Freeworld |
11:41AM |
2 |
Can asterisk be programmed to make "alarm calls"? |
11:34AM |
0 |
ERROR[147466]: chan_capi.c:1914 |
11:25AM |
0 |
ISDN & Voicemail: Strange Behaviour |
10:44AM |
1 |
poll vs select in channel.c |
10:41AM |
1 |
asterisk-doc Conference Call - phase 2 :) |
8:28AM |
0 |
IVR with chan_h323 module |
8:22AM |
0 |
T100P Hang up detection |
8:15AM |
2 |
Unable to play dialtone on channel xx (Zaptel TE405P) |
7:57AM |
4 |
BGM Music |
7:48AM |
5 |
Budgetone iLBC to IAX2 iLBC |
4:39AM |
0 |
Taiwan calling!!!! |
3:07AM |
1 |
DASS2 support |
2:55AM |
1 |
Where are the list archives?? |
2:20AM |
0 |
Consultive Transfer, or faking it |
1:44AM |
0 |
extensions in mysql |
1:41AM |
0 |
zaptel does not compile on latest RHEL kernel |
1:22AM |
1 |
SNOM200 + 2.05a Firmware + G729b BUG |
12:09AM |
0 |
MGCP channel problem |
|
Wednesday May 12 2004 |
Time | Replies | Subject |
9:47PM |
2 |
Sonicwall with Firmware 6.6.02 - SIP? |
9:07PM |
4 |
Losing my PRI Interface every 20-30 minutes??? |
7:25PM |
2 |
Newbie voiceplus + asterisk |
7:23PM |
2 |
iax behind a SonicWall |
3:58PM |
2 |
problems with analog interface to PBX |
3:25PM |
1 |
CISCO 30 VIP phone / 12 SP+ Connection does not free up |
3:21PM |
5 |
2.05a firmware |
2:59PM |
0 |
User Instructions for voicemail system |
1:33PM |
0 |
Simulating national-/internationalprefix ? |
1:04PM |
0 |
(no subject) |
12:54PM |
2 |
Simulating Dialtone ? |
12:16PM |
6 |
Dell server for asterisk question! |
11:06AM |
0 |
cisco 7960 line and extension apperance |
11:05AM |
1 |
Musical interruptions |
10:18AM |
1 |
G729 Segmentation fault |
10:12AM |
1 |
Asterisk Questions |
9:54AM |
2 |
847 IAX Provider?? |
9:46AM |
0 |
Zaphfc ver 0.0.2 |
9:39AM |
2 |
* and sip proxy auth |
9:24AM |
0 |
Chan_Capi & Modem/ttyI |
8:53AM |
0 |
quadBRI telco part hungs |
8:42AM |
3 |
Cisco 7960 SIP - DND soft key toggle? |
8:03AM |
1 |
Ring Tone - SIP / IAX2 |
7:26AM |
1 |
Asterisk not loading data into table using cdr_odbc |
7:13AM |
2 |
cdr_mysql - would index slow down? |
6:43AM |
2 |
CDR-MySQL |
6:37AM |
0 |
Problems Retrieving Voicemail Remotely |
5:08AM |
3 |
Needed Open Ports |
5:04AM |
6 |
Where to get 48 volt Power Supplies for Cisco IP Phones |
4:43AM |
2 |
Calling CHRIS BARNET (PRI / E100P / ntl) |
4:26AM |
0 |
chan_CAPI uses Modem/ttyIx ? |
3:42AM |
0 |
[DTMF] Audio-Before-Answer issues |
3:39AM |
2 |
Re: [Asterisk-doc] Conference hosting request for asterisk-doc |
3:19AM |
0 |
fine mode receive fax problem |
2:57AM |
0 |
no sound for inbound calls |
12:50AM |
0 |
Asterisk Downunder (Australia & New Zealand) |
12:41AM |
0 |
SIP using h323 to gnugk |
12:41AM |
0 |
Voice Flow |
12:18AM |
1 |
Multiple ISDN controllers & Capi |
12:04AM |
2 |
Good source for Polycom IP Phones |
|
Tuesday May 11 2004 |
Time | Replies | Subject |
11:49PM |
1 |
Fax detection on IAX2 channel? |
3:17PM |
2 |
SDP messages relating to rtpmap Question |
3:02PM |
1 |
Caller-ID for alphanumeric SIP uris |
1:21PM |
0 |
midget packet received |
1:07PM |
0 |
Disabling agent call logging |
12:27PM |
1 |
Areski CDR graph incorrect |
11:55AM |
1 |
Need help: X100P connection/configuration in GERMANY |
11:03AM |
1 |
KIRK System 600 IP and Asterisk |
9:55AM |
1 |
Kernel Freezes with T100P |
9:42AM |
2 |
Asterisk + VoiceWorks |
9:22AM |
1 |
Sipura, Asterisk, *0, and Call Waiting |
8:09AM |
1 |
Long delays when talking betwwen SIP phones |
7:58AM |
0 |
New patch for Bug 1420 |
4:10AM |
2 |
Sipgate to regular phones |
3:15AM |
3 |
Line appearances |
3:07AM |
1 |
Asterisk resource consumption.. |
2:52AM |
1 |
Use buttons (other than #) after call is bridged? |
|
Monday May 10 2004 |
Time | Replies | Subject |
10:52PM |
1 |
AGI.pm wait_for_digit() not working for me!!! |
9:26PM |
2 |
Notice for Gentoo Users In Regard to mpg123 |
9:02PM |
0 |
+5 seconds delay when Switching to macros |
8:23PM |
0 |
How do I catch someone pressing the * key? |
7:43PM |
1 |
Terrible TICKING sound |
7:08PM |
1 |
help-listening to my mailbox |
6:37PM |
1 |
ztcfg and Aastra 390 phones |
6:06PM |
0 |
CISCO 30 VIP and 12 SP+ |
4:58PM |
1 |
DNS load-balancing & SRV records |
4:47PM |
0 |
polycom ip 500 registration problems |
4:11PM |
2 |
Asterisk on a dual processor machine |
2:02PM |
1 |
Callerid via PRI |
1:16PM |
6 |
SIP calls-per-second performance test tool |
1:13PM |
5 |
Swap partition/file and Asterisk. |
12:33PM |
0 |
Uniden UIP200 Review (Repost) |
12:11PM |
2 |
alternative FXO gateway to Mediatrix 1204? |
11:22AM |
3 |
Question about Asterisk and its use |
10:53AM |
6 |
Virbiage FT201 IAX Hard Phone |
10:12AM |
0 |
Re: bill |
8:20AM |
1 |
Dropped calles (with mp3) |
8:10AM |
1 |
News |
5:58AM |
1 |
mailbox numbers |
5:53AM |
2 |
Explain cidinternalcontexts? |
5:47AM |
1 |
Signalling C7 / SS7 |
3:44AM |
1 |
app_sms - rocks! |
3:27AM |
3 |
Asterisk & Rhetorical Systems |
3:27AM |
2 |
problems compiling oh_323 and asterisk |
2:39AM |
0 |
SIP seeding |
2:02AM |
1 |
Problem with SMP? |
1:53AM |
1 |
Testing IP phone (g729, g711) with Windows Messenger (g723, g711) |
1:33AM |
0 |
SIP Error: Network Unreachable |
|
Sunday May 9 2004 |
Time | Replies | Subject |
11:22PM |
11 |
SIP in the UK |
10:56PM |
1 |
Example: calling card using extension logic ONLY! |
8:54PM |
1 |
Problems when upgraded |
5:18PM |
4 |
Asterisk webmin |
3:51PM |
2 |
Help with initial setup |
2:28PM |
3 |
German sound files available |
2:09PM |
3 |
AGI Assistance |
12:50PM |
0 |
Cannot Dial out with xp100 |
12:47PM |
1 |
AGI Assitance |
11:23AM |
3 |
DTMF broken |
11:19AM |
0 |
NOT USING REPLY TO THE LIST |
9:47AM |
1 |
No outbound calls at a PRI possible |
9:39AM |
1 |
Stripping numbers at the end of a dial pattern => extension |
7:32AM |
2 |
Sip to PSTN Gateway Configs |
7:02AM |
2 |
Help!! Music On Hold |
6:20AM |
2 |
ztdummy problem?!? |
5:59AM |
3 |
Where to start? |
3:54AM |
0 |
Telekom ISDN CFU is it possible ? |
3:48AM |
0 |
asterisk/can_capi took ISDN B Channels busy. |
1:20AM |
0 |
Re: Question |
12:59AM |
1 |
*** Asterisk sunday news: Read the sample configs, Luke! |
|
Saturday May 8 2004 |
Time | Replies | Subject |
11:12PM |
1 |
500ms usleep in rtp.c ? |
3:03PM |
1 |
Stripping numbers at the end of a dial pattern => extensions.conf |
3:02PM |
3 |
asterisk with german SIPGATE ? |
2:10PM |
2 |
x100p / Answer-> Flash -> Dial |
1:51PM |
0 |
H323 - Gatekeeper - asterisk - SIP config problems |
1:10PM |
5 |
1800 Provider |
12:08PM |
0 |
Failover Scenario - synchronizing voicemail & key files |
11:09AM |
3 |
Transfering with Grandstream Phones |
9:46AM |
2 |
List of online sip users |
9:02AM |
0 |
AVM B1 ISDN Call forwarding |
7:41AM |
0 |
need working loopstart config - t100p |
3:20AM |
0 |
authorise with h323 client at the * via gatekeeper |
3:15AM |
0 |
Indication Busy to a ZAP ISDN channel |
|
Friday May 7 2004 |
Time | Replies | Subject |
7:45PM |
0 |
Callwaiting callerid on 390s? |
7:20PM |
6 |
X100P keeping PSTN line Offhook |
5:57PM |
1 |
meetme conf-background.agi |
3:57PM |
3 |
MPG123 errors |
3:38PM |
0 |
RE: PRI, multi D channels and conventional PBXs (brian) |
2:55PM |
4 |
Concept for line appearances and bridging: anyone? |
2:43PM |
1 |
Voicemail: upgraded? |
2:17PM |
1 |
Uniden UIP200 Review |
1:34PM |
0 |
Re: [Asterisk-cvs] asterisk/apps app_dial.c,1.64,1.65 |
1:13PM |
5 |
729 licence on scsi |
12:48PM |
0 |
Manager, command action |
12:03PM |
2 |
Availability of T400P and E400P |
11:53AM |
1 |
cannot play sound files |
11:52AM |
0 |
- RE: Missing digits on TDM400P incomplete dial string - DTMF problem? - Email found in subject |
11:50AM |
0 |
Routing by called interface - Email found in subject |
11:49AM |
0 |
- Re: Routing by called interface - Email found in subject |
10:56AM |
0 |
CAPI & Gain |
10:34AM |
3 |
Routing by called interface |
10:12AM |
2 |
PRI, multi D channels and conventional PBXs |
9:01AM |
4 |
Cisco 7940 Phones as paging system? |
8:08AM |
0 |
DTMF Echo on H323 Channel |
8:08AM |
1 |
Trunk with CIRPAK |
7:45AM |
2 |
Newbie x100p install question |
7:43AM |
7 |
Asterisk and Cisco 7960 problems persist (for me, anyway) |
7:27AM |
0 |
caller id detection |
7:25AM |
1 |
Missing digits on TDM400P incomplete dial string - Email found in subject |
6:14AM |
4 |
SIP Wokflow diagram |
5:59AM |
7 |
WI FI IP phones?? |
5:48AM |
2 |
zaptel.conf question |
5:41AM |
5 |
SIP: Trouble with "Moved temporarily" (302) |
4:58AM |
0 |
PSTN line tests |
4:37AM |
0 |
modem (56k) call to PSTN |
4:01AM |
1 |
Cisco 7940 microphone volume |
3:52AM |
2 |
Trouble compiling latest CVS |
1:51AM |
2 |
quadBRI & ISDN telephone |
|
Thursday May 6 2004 |
Time | Replies | Subject |
10:33PM |
1 |
sip + zap problem |
7:09PM |
0 |
AVM Fritz doesn't pick up call or make calls |
4:52PM |
3 |
mpg123 versions ? |
4:03PM |
0 |
Please help the new guy (the s extension) |
3:07PM |
6 |
HOW TO PROGRAM NEW MODULES |
2:09PM |
1 |
polycom dialplan |
10:27AM |
0 |
Unable to find the source of the error: bad file descriptora |
9:52AM |
4 |
asterisk-oh323, new version 0.6.1 |
9:47AM |
0 |
error on loading... |
9:11AM |
5 |
Fehler beim starten... |
9:02AM |
4 |
Playing GSM files in Windows |
8:05AM |
0 |
Syntax (2) |
7:49AM |
4 |
Cisco 7920 Image |
6:31AM |
2 |
Sending Tones after * connects |
6:12AM |
0 |
no incoming audio on outgoing sipcalls |
4:56AM |
3 |
Dial internal phones problem - zaphfc |
4:46AM |
1 |
Date time problems |
2:46AM |
7 |
sip traffic. |
2:43AM |
0 |
Problem in extensions.conf |
|
Wednesday May 5 2004 |
Time | Replies | Subject |
8:27PM |
0 |
CDR reporting GUI |
8:27PM |
3 |
Mediatrix 1204 (4x FXO) |
3:46PM |
0 |
(no subject) |
3:03PM |
3 |
sip.conf and SIP client host= not recognized in some cases |
2:39PM |
3 |
sip via tcp |
1:52PM |
8 |
No Audio from Hard Phone to SIP |
1:44PM |
0 |
Asunto: Re: Syntax |
12:46PM |
0 |
Lost my G.729 licenses |
12:39PM |
2 |
chan_sip and Digest realm |
11:26AM |
3 |
Problem with PRI and overlapped dialing |
11:17AM |
9 |
Correct format of sip registry |
11:16AM |
1 |
MySQL and VoiceMail again |
10:37AM |
0 |
determining pass-thru mode |
10:04AM |
1 |
Digits in a different language... |
9:31AM |
1 |
strange sip behavior (looping back to my own extension vm) |
9:19AM |
0 |
app_sms woes |
8:06AM |
1 |
Asterisk devel. - Mediatrix dtmf bug solved |
7:43AM |
1 |
PRI to PRI fax pass through |
7:42AM |
0 |
CAPI & Eicon Crash Asterisk |
7:20AM |
4 |
Cisco 7905 vs Cisco 7905-G |
6:42AM |
0 |
Missing: Red alarm event in Asterisk Manager Interface |
6:20AM |
2 |
connect a sub telefon system? |
6:04AM |
0 |
Asterisk Dialogic support |
5:45AM |
0 |
I can not register via sip to iptel or sipgate. |
4:20AM |
7 |
* & ISDN-BRI-PTP & DID & ISDN4Linux does not show incoming number |
3:08AM |
1 |
SIP Pick up groups |
2:24AM |
0 |
Problem with extension.conf |
2:10AM |
1 |
Early B3 |
2:10AM |
1 |
Problem in Extension.conf |
1:45AM |
2 |
BUSY tone |
1:11AM |
2 |
183 Session in Progress |
|
Tuesday May 4 2004 |
Time | Replies | Subject |
11:01PM |
2 |
Adtran ta750 Configuration |
9:09PM |
0 |
Cli and the called number info |
9:05PM |
1 |
VoiceTronix V4PCI and Digium TDM40B together |
8:24PM |
5 |
ACD and/or CTI components for Asterisk |
8:02PM |
1 |
vonage sip url |
4:11PM |
4 |
mediatrix 1104 |
3:43PM |
0 |
A GOOD IP PHONE IAX OR SIP |
2:59PM |
3 |
g.729 - licenses and opinions |
1:24PM |
7 |
stun server |
11:45AM |
1 |
0.7.2 debs |
11:17AM |
1 |
Error when loading wcfxo |
11:13AM |
2 |
Dial zap and music on hold |
10:37AM |
2 |
Can Asterisk support R2 signaling |
10:30AM |
1 |
multiplle isdn card |
10:21AM |
6 |
DSL vs X100P |
9:42AM |
1 |
Pots Extensions |
9:32AM |
3 |
Linux IAX client |
8:30AM |
0 |
Quality differences of codecs from PRI to SIP |
8:21AM |
0 |
Sip provider group |
7:53AM |
3 |
would it be possible to... |
7:41AM |
0 |
Help on legacy hardware. |
6:51AM |
1 |
Syntax |
6:48AM |
1 |
asterisk + NEC integration |
6:41AM |
3 |
Maximum retries exceeded problem... |
5:25AM |
2 |
Max TE410P card on an Asterisk |
5:19AM |
1 |
Asterisk and windows h.323 gatekeeper calling problems... |
4:50AM |
1 |
How does Norvergence do it ? |
4:46AM |
2 |
If Then Else Statements - Outbound Dialling on ISDN using CAPI -Individual Dial out Plans using msns |
4:37AM |
1 |
How to implement & configure agents |
2:35AM |
1 |
MGCP: Current CVS works for you? |
2:10AM |
0 |
Siemens cordless phone |
1:06AM |
0 |
Asterisk - no outband DTMF with Mediatrix |
12:30AM |
1 |
Probs with oh323 driver: audio only in 1 direction |
12:15AM |
0 |
Czech sound files |
|
Monday May 3 2004 |
Time | Replies | Subject |
11:43PM |
0 |
Asterisk E1 and Cisco as5300 |
9:35PM |
3 |
RFD: With echo and other distortion, can ulaw/alaw line quality ever be good enough for faxing? |
8:59PM |
0 |
A-B ok; B-C ok; A-C Crap |
5:46PM |
1 |
How do you close a VoicePulse "Connect!" account? |
4:07PM |
0 |
Pulse dial: outbound? |
3:37PM |
2 |
Number of Digium cards in one box... |
1:42PM |
0 |
Cisco 7905 and as5300 + Asterisk |
1:05PM |
1 |
SIP Call transfer with RTP transfer as well? |
12:29PM |
4 |
How does Novergence do it ? |
10:41AM |
2 |
ISDN WAN ISDN bridge possible? |
10:26AM |
1 |
quad fxo |
10:08AM |
1 |
Problem with new sipura firmware 1.0.35a |
9:58AM |
0 |
Open Source SCGP |
8:58AM |
0 |
txfax: "Trainability test failed" |
8:28AM |
1 |
Asterisk remains in the media path |
8:04AM |
2 |
zap x100p |
7:45AM |
1 |
Start recording during call by pressingbutton sequence |
6:36AM |
3 |
Error building asterisk-0.9.0 |
6:01AM |
1 |
dialing a remote phone system and then entering an extension |
4:28AM |
2 |
Digital Line Distortion |
2:59AM |
1 |
Réf.: Re: Asterisk with UUI support ? |
12:43AM |
1 |
Asterisk & MGCP / NCS |
|
Sunday May 2 2004 |
Time | Replies | Subject |
11:06PM |
1 |
* Newbie installation advice |
8:16PM |
1 |
Re: Adit 600 FXO card (Jon Brandon) |
7:17PM |
1 |
Channel Bank - Vina T-1 Integrator |
6:02PM |
0 |
X101P problems |
5:41PM |
1 |
Adit 600 FXO card |
3:42PM |
1 |
Why don't I get a ringing sound? |
3:25PM |
1 |
module help? |
2:53PM |
1 |
Voicemail or voicemail2? |
1:15PM |
1 |
FXO line hum w/ Z-plex 10 |
1:00PM |
2 |
Cisco 12SP+ |
12:54PM |
0 |
chan_vpd patches |
8:57AM |
6 |
Simple SIP X-Lite Configuration Failing |
7:37AM |
1 |
no dial tone |
5:24AM |
2 |
Talking SIP to Vocal |
1:39AM |
4 |
iconnecthere behind NAT, strange deal |
12:32AM |
1 |
phonejack and linejack in the same system |
|
Saturday May 1 2004 |
Time | Replies | Subject |
7:15PM |
0 |
FW: clicks at beginning of call |
5:50PM |
1 |
Searching Archives (Basic SIP Configuration Problem)? |
3:28PM |
0 |
RE: [E164-discuss] RE: E164 updater Client |
2:36PM |
1 |
Grandstream Ringtones |
2:08PM |
4 |
New TDM04B 4-port FXO card problems |
12:30PM |
1 |
Grandspream & call parking |
12:18PM |
0 |
Reviewers Needed |
11:04AM |
1 |
Fax Detect problem (have consulted archives, wiki & irc) |
7:10AM |
0 |
h323.conf: multiple hosts per user? |
5:59AM |
1 |
dialing out to PSTN from SIP phones |
1:58AM |
1 |
Outbound Dialling on ISDN using CAPI - Individual Dial out Plans using msns |
1:02AM |
4 |
New ENUM service, what do you think? |
12:25AM |
0 |
Asterisk, festival, dropped calls |